修改BaseWebRTC 的配置注入

frontend/hooks/useWebRTCConnection.ts

@@ -3,7 +3,11 @@ import WebRTC_Initiator from "@/lib/webrtc_Initiator";
 import WebRTC_Recipient from "@/lib/webrtc_Recipient";
 import FileSender from "@/lib/fileSender";
 import FileReceiver from "@/lib/fileReceiver";
-import { config } from "@/app/config/environment"; // For API_URL
+import {
+  config,
+  getIceServers,
+  getSocketOptions,
+} from "@/app/config/environment";
 import type { CustomFile, fileMetadata, FileMeta } from "@/types/webrtc"; // Assuming FileMeta might be used by caller
 import type { Messages } from "@/types/messages";
 
@@ -55,8 +59,13 @@ export function useWebRTCConnection({
 
   // Initialize WebRTC objects and their cleanup
   useEffect(() => {
-    const senderConn = new WebRTC_Initiator(config.API_URL);
-    const receiverConn = new WebRTC_Recipient(config.API_URL);
+    const webRTCConfig = {
+      iceServers: getIceServers(),
+      socketOptions: getSocketOptions()|| {},
+      signalingServer: config.API_URL,
+    };
+    const senderConn = new WebRTC_Initiator(webRTCConfig);
+    const receiverConn = new WebRTC_Recipient(webRTCConfig);
     setSender(senderConn);
     setReceiver(receiverConn);
 

frontend/lib/webrtc_Initiator.ts

@@ -1,11 +1,11 @@
 // 发起方 流程: 加入房间; 收到 'ready' 事件（新的接收方进入后socker server就会触发这个事件） -> createPeerConnection + createDataChannel -> createAndSendOffer
-import BaseWebRTC from './webrtc_base';
+import BaseWebRTC, { WebRTCConfig } from './webrtc_base';
 import { postLogInDebug } from '@/app/config/api';
 const developmentEnv = process.env.NEXT_PUBLIC_development!;//开发环境
 
 export default class WebRTC_Initiator extends BaseWebRTC {
-  constructor(signalingServer: string) {
-    super(signalingServer);
+  constructor(config: WebRTCConfig) {
+    super(config);
     this.setupInitiatorSocketListeners();
 
   }

frontend/lib/webrtc_Recipient.ts

@@ -1,5 +1,5 @@
 // 接收方 流程: 加入房间; 收到 'offer' 事件 -> createPeerConnection + createDataChannel -> 发送 answer
-import BaseWebRTC from './webrtc_base';
+import BaseWebRTC, { WebRTCConfig } from './webrtc_base';
 import { postLogInDebug } from '@/app/config/api';
 const developmentEnv = process.env.NEXT_PUBLIC_development!;//开发环境
 
@@ -8,8 +8,8 @@ interface AnswerPayload {
   peerId: string;
 }
 export default class WebRTC_Recipient extends BaseWebRTC {
-  constructor(signalingServer: string) {
-    super(signalingServer);
+  constructor(config: WebRTCConfig) {
+    super(config);
     this.setupRecipientSocketListeners();
   }
 

frontend/lib/webrtc_base.ts

@@ -1,6 +1,5 @@
 // BaseWebRTC.js
-import io from 'socket.io-client';
-import { Socket } from 'socket.io-client';
+import io, { Socket, ManagerOptions, SocketOptions } from 'socket.io-client';
 import { getIceServers, getSocketOptions } from '@/app/config/environment';
 import { WakeLockManager } from './wakeLockManager';
 import { postLogInDebug } from '@/app/config/api';
@@ -19,6 +18,12 @@ interface CallbackTypes {
   onConnectionStateChange?: (state: RTCPeerConnectionState, peerId: string) => void;
 }
 
+export interface WebRTCConfig {
+  iceServers: RTCIceServer[];
+  socketOptions: Partial<ManagerOptions & SocketOptions>;
+  signalingServer: string;// signalingServer: 信令服务器的URL，用于初始化Socket.IO连接。
+}
+
 export default class BaseWebRTC {
   //类型申明
   protected iceServers: RTCIceServer[];
@@ -42,9 +47,9 @@ export default class BaseWebRTC {
   protected reconnectionInProgress: boolean;//防止重复重连
   protected wakeLockManager: WakeLockManager;
 
-  constructor(signalingServer: string) {// signalingServer: 信令服务器的URL，用于初始化Socket.IO连接。
-    this.iceServers = getIceServers();
-    this.socket = io(signalingServer, getSocketOptions());
+  constructor(config: WebRTCConfig) {
+    this.iceServers = config.iceServers;
+    this.socket = io(config.signalingServer, config.socketOptions);
     this.peerConnections = new Map();// Map<targetPeerId, RTCPeerConnection>
     this.dataChannels = new Map();// Map<targetPeerId, RTCDataChannel>